+=== release 1.1.2 ===
+
+2013-07-11 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ releasing 1.1.2
+
+2013-07-10 17:16:14 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Only give sinks a new bus if they have no parent yet
+ Otherwise we will remove the bus that would proxy messages to playsink
+ and never set it again. If the sink is already in playsink, all failures
+ are fatal anyway as it's either a sink that worked before or one that
+ was set by the user.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701997
+
+2013-07-10 13:22:04 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Store a/v/t sinks locally too, not just in playsink
+
+2013-07-10 13:21:29 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaysink.c:
+ playsink: ref_sink() any sinks that are set on playsink
+ Otherwise the behaviour of the properties is inconsistent.
+
+2013-07-10 13:20:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * tests/check/elements/playbin.c:
+ playbin: Fix assumptions in the unit test
+ Unused sinks are still set to READY now during autoplugging
+ to check their caps. Also playsink owns a ref to the sinks too.
+
+2013-07-10 13:00:21 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gststreamsynchronizer.c:
+ streamsynchronizer: Non-TIME segment streams are not waiting automatically
+ This was leftover code from porting to 1.0 and fixes the playbin
+ unit test.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701943
+
+2013-07-09 23:04:49 +0200 Branko Subasic <branko@axis.com>
+
+ * win32/common/libgstrtp.def:
+ win32: add missing rtp buffer methods
+
+2013-07-09 14:55:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ * gst/playback/gstplaysink.c:
+ playbin: Change sink ownership handling to be a bit more sane
+ playbin will now only activate the sinks in a single place and
+ will never change the states of any sinks that are owned by
+ playsink.
+ Also handle text-sinks the same way as audio/video sinks inside
+ playbin.
+
+2013-07-05 21:55:26 +0200 Piotr Drąg <piotrdrag@gmail.com>
+
+ * po/POTFILES.in:
+ po: update POTFILES.in
+ https://bugzilla.gnome.org/show_bug.cgi?id=703684
+
+2013-07-04 17:09:00 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst-libs/gst/video/colorbalance.c:
+ colorbalance: Fix the typo in base_init().
+
+2013-07-04 12:54:59 -0400 Thibault Saunier <thibault.saunier@collabora.com>
+
+ * gst/adder/gstadder.c:
+ adder: Do not send flush_start event with the stream lock taken
+ FLUSH_START is not serialized, so the lock should not be taken when
+ sending it.
+
+2013-07-05 00:47:08 +0100 Marcin Lewandowski <marcin@saepia.net>
+
+ * gst-libs/gst/tag/id3v2frames.c:
+ tag: ignore malformed ID3v2 TDAT frames
+ Just skip them, don't cause criticals.
+ https://bugzilla.gnome.org/show_bug.cgi?id=703283
+
+2013-07-03 09:44:32 +0100 Tim-Philipp Müller <tim@centricular.net>
+
+ * gst/audioresample/speex_resampler_int.c:
+ audioresample: make explicit that neon is disabled and why
+ https://bugzilla.gnome.org/show_bug.cgi?id=703477
+
+2013-07-02 18:20:39 +0200 Carlos Rafael Giani <dv@pseudoterminal.org>
+
+ * gst/audioresample/speex_resampler_int.c:
+ audioresample: disable 16-bit integer NEON support
+ it seems to be broken (produces no audio), plus the performance gain
+ is small
+ Signed-off-by: Carlos Rafael Giani <dv@pseudoterminal.org>
+
+2013-07-02 14:25:28 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: If we had a previous autoplugged sink, try to reuse it
+ https://bugzilla.gnome.org/show_bug.cgi?id=701997
+
+2013-07-02 14:18:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaysink.c:
+ playsink: If we switch sinks, make sure that the old sink is set to NULL
+
+2013-07-02 14:02:57 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Don't change the state of sinks that we passed to playsink already
+
+2013-07-02 14:01:52 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Consider new audio/video sinks when reconfiguring
+
+2013-07-02 12:27:03 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Improve debug output regarding sink selection
+
+2013-07-01 12:52:43 -0600 Brendan Long <self@brendanlong.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Post an error message if a stream combiner doesn't return a request pad.
+
+2013-07-01 13:45:25 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Only intersect to check if a sink can handle raw caps
+ Doing a subset check requires fixed caps, which we might not have here.
+ https://bugs.webkit.org/show_bug.cgi?id=116042
+
+2013-07-01 10:39:02 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * gst-libs/gst/pbutils/missing-plugins.c:
+ * gst-libs/gst/pbutils/pbutils-private.h:
+ pbutils: allow describing unfixed caps if they share the same media type
+ Caps description and missing plugin code does not really need caps to
+ be fixed, and indeed they may not be if giving encodebin unfixed caps
+ that correspond to an unknown encoder or muxer.
+ So we relax the check, and allow unfixed caps if all the structures
+ refer to the same media type.
+
+2013-07-01 11:16:34 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Send all pending events with type < CAPS before sending caps
+
+2013-06-27 16:33:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst-libs/gst/video/gstvideoencoder.c:
+ videoencoder: Send all pending events with type < CAPS before sending caps.
+ https://bugzilla.gnome.org/show_bug.cgi?id=703196
+
+2013-06-28 14:48:19 +0100 Vincent Penquerc'h <vincent.penquerch@collabora.co.uk>
+
+ * gst/typefind/gsttypefindfunctions.c:
+ typefind: avoid too low mpeg/ts probability on small amount of data
+ With the current test, we get into problems when we try to typefind
+ a MPEG stream from a small amount of data, which can happen when
+ we get data pushed from a HTTP source. We thus make a second test
+ to give higher probability if all the potential headers were either
+ pack or pes headers (ie, no potential header was unrecognized).
+ This fixes an issue with a MPEG1/MP2 stream being properly discovered
+ as video/mpeg from a file, but as audio/mpeg from souphttpsrc.
+ https://bugzilla.gnome.org/show_bug.cgi?id=703256
+
+2013-06-30 18:17:15 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ * gst-libs/gst/video/gstvideoencoder.c:
+ video(enc|dec)oder: Don't return not-negotiated if flushing
+ If the pad is flushing after a failed negotiation, return
+ GST_FLOW_FLUSHING instead from finish_frame().
+ https://bugzilla.gnome.org/show_bug.cgi?id=701763
+
+2013-06-30 18:16:35 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: Don't return not-negotiated if flushing
+ If the pad is flushing after a failed negotiation, return
+ GST_FLOW_FLUSHING instead from finish_frame().
+ https://bugzilla.gnome.org/show_bug.cgi?id=701763
+
+2013-06-14 07:23:40 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ * tests/check/libs/pbutils.c:
+ pbutils: descriptions: Allow smart codec tag handling
+ We already have internally the information on what type of stream (audio,
+ video, container, subtitle, ...) a certain caps is.
+ Instead of forcing callers to specify which CODEC_TAG category a certain
+ caps is, use that information to make a smart choice.
+ Does not break previous behaviour of gst_pb_utils_add_codec_description_to_tag_list
+ (if tag is specified it will be used, if caps is invalid it will be rejected,
+ ...).
+ https://bugzilla.gnome.org/show_bug.cgi?id=702215
+
+2013-06-19 09:25:48 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmptag: Add a debug category
+ Instead of using the default category
+
+2013-06-27 12:23:27 +0200 Patricia Muscalu <patricia@axis.com>
+
+ * gst/videotestsrc/gstvideotestsrc.c:
+ videotestsrc: do not leak lines
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703177
+
+2013-06-26 14:36:17 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbasepayload.c:
+ rtpbasepayload: Do not leak the event when segment is delayed
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703119
+
+2013-06-26 15:03:05 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: make read uncancelable when reading a message
+ When we start to read a message, we need to continue reading until the end of
+ the message or else we lose track and cause parse errors. Use a variable
+ may_cancel to avoid cancelation after we read the first byte until we have
+ the complete message.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=703088
+
+2013-06-21 20:41:15 +0200 Mathieu Duponchelle <mathieu.duponchelle@epitech.eu>
+
+ * gst-libs/gst/audio/gstaudiodecoder.c:
+ audiodecoder: Don't return not-negotiated if flushing
+ If the pad is flushing after a failed negotiation, return GST_FLOW_FLUSHING.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701763
+
+2013-06-23 12:07:41 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/ogg/gstoggstream.c:
+ ogg: The Daala headers are little endian, not big endian
+
+2013-06-23 10:30:02 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * ext/ogg/gstoggmux.c:
+ * ext/ogg/gstoggstream.c:
+ ogg: Add Daala support
+
+2013-06-21 19:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/pbutils/descriptions.c:
+ pbutils: Add VP9 description
+
+2013-06-17 08:58:13 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/video/gstvideodecoder.c:
+ videodecoder: Fix drop frame handling at startup
+ In the unlikely case that the decoder drops a frame before the first
+ input frame is outputted, use the input segment (since it wasn't
+ carried over to the output segment yet)
+ https://bugzilla.gnome.org/show_bug.cgi?id=702502
+
+2013-06-21 11:50:33 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: dispatch when initial buffer has data
+ When we have data in the inital buffer, dispath the read function to read it
+ even if the socket has no data to read.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702652
+
+2013-06-20 17:28:46 +0200 Wim Taymans <wim.taymans@collabora.co.uk>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtsp: manage writer child source better
+ Only add the write child source when we have something to write or else
+ we will dispatch forever without doing anything.
+
+2013-06-19 13:21:45 +0200 Jonas Holmberg <jonashg@axis.com>
+
+ * gst-libs/gst/audio/gstaudioencoder.c:
+ audioencoder: unref before memset
+ Unref allocator and input_caps in encoder context before memsetting the
+ context.
+
+2013-06-19 09:22:50 +0200 Edward Hervey <edward@collabora.com>
+
+ * gst-libs/gst/tag/gstxmptag.c:
+ xmptag: More efficient GSList usage
+ Instead of constantly appending (which gets more and more expensive), just
+ prepend to the list (O(1)) and reverse the list before usage.
+ https://bugzilla.gnome.org/show_bug.cgi?id=702545
+
+2013-06-16 22:39:30 +0200 Branko Subasic <branko@axis.com>
+
+ * gst-libs/gst/rtp/gstrtpbuffer.c:
+ * gst-libs/gst/rtp/gstrtpbuffer.h:
+ * tests/check/libs/rtp.c:
+ rtpbuffer: add gst_rtp_buffer_get_payload_bytes
+ The function gst_rtp_buffer_get_payload can not be used in Python
+ because it lacks necessary length parameter. This patch adds a new
+ function, gst_rtp_buffer_get_payload_bytes, to use from Python
+ bindings. The new function has the advisory "Rename to:" annotation
+ so it can replace the gst_rtp_buffer_get_payload whan creating
+ bindings.
+ The function gst_rtp_buffer_get_extension_bytes is also added. It wraps
+ gst_rtp_buffer_get_extension_data which doesn't work in Python due to
+ incomplete annotation and because it returns the length as number of
+ 32-bit words.
+ https://bugzilla.gnome.org/show_bug.cgi?id=698562
+
+2013-06-17 16:34:26 +0200 Ognyan Tonchev <ognyan@axis.com>
+
+ * gst-libs/gst/audio/gstaudiobasesrc.c:
+ audiobasesrc: add 2 missing gst_buffer_unmap () calls
+ There are 2 missing calls to gst_buffer_unmap () in the error handling in
+ create ().
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702467
+
+2013-06-17 16:02:41 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * gst/playback/gstplaysink.c:
+ playsink: Fix the block diagram of deinterlace bin.
+ Fixes https://bugzilla.gnome.org/show_bug.cgi?id=702465
+
+2013-06-13 11:08:20 -0600 Brendan Long <b.long@cablelabs.com>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Emit {audio,text,video}-changed signals when pads are removed
+ https://bugzilla.gnome.org/show_bug.cgi?id=702195
+
+2013-06-11 15:22:50 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/videoconvert/videoconvert.c:
+ videoconvert: Fix leaking of the chroma resample helper objects
+
+2013-06-10 14:43:35 +0300 Sreerenj Balachandran <sreerenj.balachandran@intel.com>
+
+ * tests/check/Makefile.am:
+ * tests/check/elements/playbin-complex.c:
+ tests: add more unit test for playbin
+ Add unit test for autoplugging of video_decoder/video_sink combination
+ based on capsfeatures.
+
+2013-06-10 15:31:38 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst-libs/gst/rtsp/gstrtspconnection.c:
+ rtspconnection: Make sure to set a sensible default port for the GSocketConnection
+ Otherwise it will connect to port 0 if no port is given in the URI.
+ https://bugzilla.gnome.org/show_bug.cgi?id=701798
+
+2013-06-09 19:20:20 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/adder/gstadder.c:
+ adder: Reject segments that have a different rate than the output segment
+ adder does no rate conversion.
+
+2013-06-08 23:51:13 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: When activating a fixed sink, proxy error messages too
+ If activating a fixed sink fails, everything will fail later anyway
+ and we can just error out early.
+
+2013-06-08 23:34:53 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Improve autoplugging of decoder/sink combinations by trying to activate the sink
+ And if that fails don't bother autoplugging that sink. Also gives
+ us more accurate sink caps.
+
+2013-06-08 23:08:05 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Proxy the playbin context to the sinks
+
+2013-06-08 23:04:43 +0200 Sebastian Dröge <slomo@circular-chaos.org>
+
+ * gst/playback/gstplaybin2.c:
+ playbin: Proxy sink messages if we activate a sink in playbin already
+ This makes sure the application gets any context related messages and
+ can do whatever is required to a) get the sink a context or b) share
+ the context with other elements in the pipeline.
+ The proxying is necessary because the sink is not a child element of
+ playbin, but instead will at a later point be a child of some bin
+ inside playsink.
+ https://bugzilla.gnome.org/show_bug.cgi?id=700967
+
+2013-06-06 15:57:49 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * gst/playback/gstdecodebin2.c:
+ decodebin: Let serialize queries before caps events through
+ Otherwise we're going to deadlock forever because no autoplugging
+ happens without having caps, but caps can never be send because
+ we're blocking.
+ Serialized queries before caps should never be sent unless really
+ necessary.
+
+2013-06-05 18:36:40 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+
+ * configure.ac:
+ Back to development
+
=== release 1.1.1 ===
-2013-06-05 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+2013-06-05 17:58:51 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
+ * ChangeLog:
+ * NEWS:
+ * RELEASE:
+ * common:
* configure.ac:
- releasing 1.1.1
+ * docs/plugins/gst-plugins-base-plugins.args:
+ * docs/plugins/gst-plugins-base-plugins.hierarchy:
+ * docs/plugins/gst-plugins-base-plugins.interfaces:
+ * docs/plugins/gst-plugins-base-plugins.signals:
+ * docs/plugins/inspect/plugin-adder.xml:
+ * docs/plugins/inspect/plugin-alsa.xml:
+ * docs/plugins/inspect/plugin-app.xml:
+ * docs/plugins/inspect/plugin-audioconvert.xml:
+ * docs/plugins/inspect/plugin-audiorate.xml:
+ * docs/plugins/inspect/plugin-audioresample.xml:
+ * docs/plugins/inspect/plugin-audiotestsrc.xml:
+ * docs/plugins/inspect/plugin-cdparanoia.xml:
+ * docs/plugins/inspect/plugin-encoding.xml:
+ * docs/plugins/inspect/plugin-gio.xml:
+ * docs/plugins/inspect/plugin-ivorbisdec.xml:
+ * docs/plugins/inspect/plugin-libvisual.xml:
+ * docs/plugins/inspect/plugin-ogg.xml:
+ * docs/plugins/inspect/plugin-pango.xml:
+ * docs/plugins/inspect/plugin-playback.xml:
+ * docs/plugins/inspect/plugin-subparse.xml:
+ * docs/plugins/inspect/plugin-tcp.xml:
+ * docs/plugins/inspect/plugin-theora.xml:
+ * docs/plugins/inspect/plugin-typefindfunctions.xml:
+ * docs/plugins/inspect/plugin-videoconvert.xml:
+ * docs/plugins/inspect/plugin-videorate.xml:
+ * docs/plugins/inspect/plugin-videoscale.xml:
+ * docs/plugins/inspect/plugin-videotestsrc.xml:
+ * docs/plugins/inspect/plugin-volume.xml:
+ * docs/plugins/inspect/plugin-vorbis.xml:
+ * docs/plugins/inspect/plugin-ximagesink.xml:
+ * docs/plugins/inspect/plugin-xvimagesink.xml:
+ * gst-libs/gst/audio/gstaudiopack-dist.c:
+ * gst-libs/gst/video/video-orc-dist.c:
+ * gst-libs/gst/video/video-orc-dist.h:
+ * gst-plugins-base.doap:
+ * gst/audioconvert/gstaudioconvertorc-dist.c:
+ * gst/videoconvert/gstvideoconvertorc-dist.c:
+ * gst/videoscale/gstvideoscaleorc-dist.c:
+ * gst/volume/gstvolumeorc-dist.c:
+ * po/af.po:
+ * po/az.po:
+ * po/bg.po:
+ * po/ca.po:
+ * po/cs.po:
+ * po/da.po:
+ * po/de.po:
+ * po/el.po:
+ * po/en_GB.po:
+ * po/eo.po:
+ * po/es.po:
+ * po/eu.po:
+ * po/fi.po:
+ * po/fr.po:
+ * po/gl.po:
+ * po/hu.po:
+ * po/id.po:
+ * po/it.po:
+ * po/ja.po:
+ * po/lt.po:
+ * po/lv.po:
+ * po/nb.po:
+ * po/nl.po:
+ * po/or.po:
+ * po/pl.po:
+ * po/pt_BR.po:
+ * po/ro.po:
+ * po/ru.po:
+ * po/sk.po:
+ * po/sl.po:
+ * po/sq.po:
+ * po/sr.po:
+ * po/sv.po:
+ * po/tr.po:
+ * po/uk.po:
+ * po/vi.po:
+ * po/zh_CN.po:
+ * win32/common/_stdint.h:
+ * win32/common/audio-enumtypes.c:
+ * win32/common/config.h:
+ * win32/common/video-enumtypes.c:
+ * win32/common/video-enumtypes.h:
+ Release 1.1.1
2013-06-05 16:20:38 +0200 Sebastian Dröge <sebastian.droege@collabora.co.uk>
-Release notes for GStreamer Base Plugins 1.1.1
+Release notes for GStreamer Base Plugins 1.1.2
The GStreamer team is proud to announce a new bug-fix release
Bugs fixed in this release
- * 700342 : decodebin: Crashes and deadlocks when setting to READY while still autoplugging
- * 690197 : alsasrc: gets stuck in infinite loop if usb audio device is disconnected while being used
- * 697112 : GLTextureUploadMeta: No support for multi-texture formats
- * 634407 : decodebin should expose pads in a deterministic order
- * 636753 : pbutils function to map (container) caps to filename extension
- * 654830 : discoverer, uridecodebin, encodebin and multiple audio streams
- * 663350 : theoraenc: do not reset the encoder when we need a keyframe
- * 665751 : video: define for formats supported by gst_video_overlay_composition_blend()
- * 676884 : audiotestsrc: segment one sample too short due to rounding errors
- * 678892 : uridecodebin: differentiate between no URI handler found and URI not accepted by handler
- * 679456 : videodecoder: fix compiler optimization hint macro usage
- * 681719 : audiovisualizer does not handle VideoMeta
- * 685637 : [audioresample] Performance improvements & ARM NEON support
- * 687146 : rtpbasedepay: remove unused variable
- * 687284 : audioconvert: prefer output formats with the same depth or at least a higher depth
- * 687466 : audiobasesink: use the same type as the internal type to return it
- * 687472 : video-blend: fix memory leak
- * 687817 : textoverlay: support shaded background drawing for all formats
- * 689326 : multifdsink: document that adding fd in NULL is not allowed
- * 689845 : Encodebin API to handle multiple streams lacking
- * 690240 : encodebin: remove test of encoder name vs preset name
- * 690591 : No decoder available for type 'audio/x-avi-unknown, codec_id=(int)65534'.
- * 690994 : videodecoder: Allow parse function to not use all data on the adapter
- * 691072 : decodebin: Doesn't expose pads if no data is received before EOS
- * 692358 : appsrc deadlock setting the pipeline to NULL state
- * 692613 : tests: reduce number of wake-ups in test applications
- * 692930 : avidemux: add raw 8-bit monochrome
- * 693302 : decodebin: g_mutex_new is deprecated
- * 693401 : gstdecodebin2 doesn't set send event on pad before exposing pad
- * 693484 : uridecodebin: query URI to source element and fallback to decoder's URI
- * 693750 : Riffmedia doesn't set systemstream=false for some video/mpeg caps
- * 693862 : Crash in videoscale (with Orc enabled) on Raspberry Pi
- * 694346 : pbutils, typefinding: improve handling of MVC/SVC H.264 streams
- * 694389 : non flushing seeks after a segment done, don't sync the ringbuffer
- * 694443 : libgstaudio: add support for AAC pass-through
- * 694553 : adder: rhythmbox crossfading stopped working after commit a86ca53
- * 695203 : xvimagesink: crash in gst_xvimagesink_xvimage_put() with HLS bip-bop stream after a while
- * 695276 : libsabi test needs an update for i386
- * 695540 : riff: support raw avi with negative height
- * 695658 : build: Link libgstrtsp-1.0.so to libm for pow()
- * 695660 : appsink: update the emit-signal description
- * 695832 : audio: a print causes a floating point exception
- * 696100 : videoconvert/videoscale: broken conversion for interlaced Y41B
- * 696411 : audiotestsrc: incorrect data size in last buffer
- * 696550 : riff: add " note " tag
- * 696598 : decodebin pads no longer match order in file
- * 696818 : rtsprange: use gst_util_gdouble_to_guint64 in get_seconds
- * 696915 : decodebin: get_sticky event STREAM_START fails on newly-exposed pad
- * 696916 : videofilter doesn't add caps in pool config
- * 697628 : ximagesink: Compile error without HAVE_XSHM
- * 697631 : videoscale and videoconvert unit tests need to be updated for latest changes
- * 697665 : Add format=WMV3 for WMV 3 video
- * 697672 : VP8 passed through rtpbin decodes a single frame and then fails to decode until a key frame passed through
- * 697723 : audioringbuffer: Reset segdone when releasing audioringbuffer
- * 697808 : sdp: add boxed type for GstSDPMessage
- * 698277 : Use gst_plugin_feature_rank_compare() API instead of duplicating the code in many places
- * 698410 : Adder: Can not send flush_start and flush_stop in a row
- * 698558 : sdp: make it possible to modify session/media attributes
- * 698712 : playbin: autoplug video decoder and sink based on caps features
- * 698851 : playbin: ability to mix or play multiple audio and text streams simultaneously
- * 698888 : SDP session bandwidth not duplicated, causing segfault when freeing...
- * 699124 : vorbisdec: crash on shutdown in webkit unit test
- * 699187 : videorate: ends up outputting buffers with incorrect duration
- * 699470 : dmabuf: handle mmap failure
- * 699563 : dmabuf: fix formating
- * 699565 : dmabuf: fix memory initialization
- * 699566 : dmabuf: don't touch the GstMemory size
- * 699744 : alsasrc: timestamps provided by audiosrc subclass not used when running under slave clock
- * 699792 : oggmux: Never emitting EOS in GES
- * 699894 : videoencoder: Caps event sent before stream-start
- * 699960 : videodecoder: Reordering sticky events
- * 699971 : oggmux: Sends a segment event before sending a caps event.
- * 700006 : audio/video: base classes have suboptimal error handling when allocating a buffer not via a bufferpool
- * 700222 : rtpbasepayload: Need to delay segments event after caps event
- * 700259 : audio: fix buffer overflow for channels > 64
- * 700272 : playback: Use subset checks instead of intersections
- * 700324 : playbin hangs trying to play 4K video, and hangs again on interrupt
- * 700377 : video: add NV16 pixel format support
- * 700400 : video: can't build without orc support - implicit declaration of function 'video_orc_pack_NV16'
- * 700411 : dmabuf: Make sure that memory is unmapped before releasing it
- * 700413 : ximagesink: add alpha mask support
- * 700427 : dmabuf: set the initial memory size to the full size
- * 701202 : playsink: Badly initialized contrast/brightness
- * 701234 : SIGSEGV in videoconvert_convert_free when using fastpath
- * 701316 : rtspconnection: using g_pollable_stream_read and write breaks builds on Ubuntu and Debian stable
- * 589242 : videoconvert: need special handling for interlaced I420
- * 648359 : baseaudiosrc: ringbuffer: segbase/segdone not updated when ring buffer cleared leads to incorrect timestamps
+ * 696300 : H264 video is playing too fast because of invalid PTS:
+ * 698562 : rtpbuffer: broken language bindings for gst_rtp_buffer_get_payload
+ * 700967 : playbin: sink messages are not received in the bus
+ * 701798 : rtspsrc: Regression with connections to certain live stream
+ * 701924 : tests: add playbin test for autoplugging of decoder-sink combination based on capsfeatures
+ * 701943 : playbin: unit tests fail
+ * 701997 : [regression] playbin: loses audio clock and hangs when switching songs after about-to-finish
+ * 702195 : playbin: Emit {audio,text,video}-changed signals when pads are removed
+ * 702215 : pbutils: descriptions: Allow smart codec tag handling
+ * 702465 : playsink: Fix the block diagram of deinterlace bin.
+ * 702467 : audiobasesrc: missing calls to gst_buffer_unmap
+ * 702545 : tags: xmp: adding tags is very slow
+ * 702652 : rtspconnection: tunneled connections do not work
+ * 702840 : queue2 does not work with use-buffering=true
+ * 703088 : rtspsrc parse error race condition
+ * 703119 : rtpbasepayload leaks the segment event
+ * 703196 : videoencoder: Send all pending events with type < CAPS before sending caps.
+ * 703256 : typefind: Fix low probability typefinding for video/mpeg on small amount of data
+ * 703477 : audioresample: disable NEON code path for 16-bit integer
+ * 703684 : POTFILES.in is out of date
==== Download ====
Contributors to this release
- * Akihiro Tsukada
- * Alessandro Decina
- * Alexander Schrab
- * Andoni Morales Alastruey
- * Anton Belka
- * Arnaud Vrac
- * B.Prathibha
- * Benjamin Gaignard
+ * Branko Subasic
* Brendan Long
* Carlos Rafael Giani
- * Christian Fredrik Kalager Schaller
- * Daniel Drake
- * David Schleef
- * David Svensson Fors
- * Dirk Van Haerenborgh
* Edward Hervey
- * Emanuele Aina
- * Evan Nemerson
- * Greg Rutz
- * Jan Schmidt
- * Jan Schole
- * Jihyun Cho
* Jonas Holmberg
- * Jonathan Liu
- * Jose Antonio Santos Cadenas
- * Josep Torra
- * Julien Moutte
- * Marc Leeman
- * Martin Pitt
- * Matej Knopp
+ * Marcin Lewandowski
* Mathieu Duponchelle
- * Matthew Waters
- * Michael Olbrich
- * Miguel Angel Cabrera Moya
- * Nicola Murino
- * Nicolas Dufresne
* Ognyan Tonchev
- * Olivier Crête
* Patricia Muscalu
- * Paul HENRYS
- * Pete Beardmore
- * Philippe Normand
- * Rasmus Rohde
- * Rico Tzschichholz
+ * Piotr Drąg
* Sebastian Dröge
- * Sebastian Rasmussen
- * Simon Berg
* Sreerenj Balachandran
- * Stefan Sauer
- * Thiago Santos
* Thibault Saunier
- * Thijs Vermeir
- * Thomas Scheuermann
* Tim-Philipp Müller
- * Tom Greenwood
* Vincent Penquerc'h
- * Víctor Manuel Jáquez Leal
* Wim Taymans
- * yanghuolin
\ No newline at end of file
projects / platform / upstream / gstreamer.git / commitdiff